The present invention is related to a method for improving handover in reception and transmission in a digital cellular mobile communication system, which includes base stations and mobile stations, in which a mobile station receives from a base station a handover command during the reception or transmission of speech on a first reception or transmission channel, whereby in the handover the reception/transmission is changed to a second reception/transmission channel. Particularly the invention is related to mobile systems based on time division multiple access, in which transmission and reception occur in bursts in separate time slots, and it is intended to be used in a mobile station.
Mobile systems in use today are mainly cellular systems, in which the system network comprises adjacent cells C, which all include a base station BS. Each of the cells thus comprises the coverage area of one base station. In addition, the cellular system also includes mobile stations MS and mobile telephone exchanges MSC which monitor several cells C and the traffic and operation of the system. The structure of the cellular system is depicted in FIG. 1.
To illustrate the invention and its background the operation of a cellular system and a mobile station will be described below using as an example the digital US TDMA mobile system based on time division multiple access and used in the United States. The system is described with reference to FIGS. 2a, 2b, 3, 4a and 4b, of which FIGS. 2a and 2b show the structure of the transmission and reception frames used in the system, FIG. 3 is a generalized block diagram of a digital mobile phone, and FIGS. 4a and 4b show the structure of time slots used in the US TDMA system.
In a digital system, information is processed using speech encoding and channel encoding. In a fixed digital telephone network, speech is transmitted as a digital signal which is PCM-coded (PCM =Pulse Code Modulation) to 64 kbit/s transmission rate. To transmit information at this speed in the radio network would require a broad channel bandwidth, so to fit the information in a narrow 30 kHz channel, the 64 kbit/s transmission rate is slowed down by means of speech encoding to 7950 bit/s.
In a digital mobile phone according to FIG. 3, transmission and reception occur as follows. In the first stage of the transmission sequence, analog speech is digitized 1 and encoded 2. Sampling at, say, 64 kbit/s is performed with an A/D converter 1, which may also be called a PCM-speech encoder 1, and the samples are segmented into 20-ms speech frames comprising 1280 bits (64,000 bit/s * 20 ms=1280 bit). 20 ms of analog speech is sampled with the PCM-speech encoder I using a kind of buffer into which it is collected one 20-ms speech frame at a time. When a 20-ms speech frame has been collected, it is taken to a speech encoder 2 which removes the recurrence in the speech signal waveform and characteristics which are not essential for the intelligibility of the signal. The input signal to the speech encoder 2 is updated at 20-ms intervals. Various known algorithms, e.g. VSELP (Vector Sum Excited Linear Predictive) coding, can be used for speech encoding 2. The encoding operations are carried out frame by frame. From the speech encoder 2 the bits are taken to the output at a transmission rate of 7950 bit/s, so 20 ms of speech at the speech encoder output comprises 159 bits (7950 bit/s * 20 ms=159 bit).
On the transmission path, noise is added to the signal and it becomes distorted. With channel encoding, both the speech and the signalling information are protected against disturbances on the radio channel. After the speech encoding 2, channel encoding 3 is performed in two stages, whereby first some of the 159 bits (12 most significant bits) are protected with block code 3a (=CRC, Cyclic Redundancy Check) and then these and the next significant bits are further protected with convolution code 3b (1/2 -rate convolution coding), and some of the bits are taken unprotected. Error correction protecting against random errors is based on comparing redundant bits produced by the convolution codec 3b. The half-rate convolution coding doubles the number of bits, thereby increasing the redundancy of the speech frame. Of the speech information only certain bits are convolution coded 3b to minimize the bandwidth requirement. To facilitate error detection, a CRC generator 3a produces parity bits from the twelve most significant bits by dividing by a certain polynomial formula. The remainder bits constitute the CRC parity bits. If dividing the received bits by the same polynomial formula as in the transmission does not result in parity bits identical with the received parity bits, then the receiver (channel decoder 15 and control unit 19) will know that an error has occurred in the information. When an error is detected, the information frame is corrected or rejected. As a result of codings 2, 3a, 3b, one frame comprises 260 bits.
As shown in FIG. 3, signalling and logic messages come directly from the control unit 19, which controls the blocks of the telephone, to the block coding block 3a and, therefore, these data messages are naturally not speech encoded 2. Similarly, signalling and logic messages received in the reception, such as the so-called call paging messages sent to a mobile station by a base station, are taken from the channel decoding block 15 to the control unit 19.
So, all in all it is sent 260 bits per one 20-ms speech frame. Deep fadings on the radio path may distort several bits in one burst and result in the rejection of the whole frame. The effect of burst errors is diminished by interleaving 5. Thus, the 260 bits of a frame are interleaved, thereby spreading the data stream (message) into several time slots. Transmission errors usually occur as error bursts, and inter-leaving aims to spread them evenly across the transmitted data, thereby optimizing the channel decoding.
A burst to be transmitted, shown in FIG. 4a, is generated 6 by adding to the inter-leaved data a training sequence, tail bits and guard time. Advantageously, the burst generation block 6 includes a buffer register into which the transmission burst is collected before transmission.
In the US TDMA system used in the United States, a digital traffic channel implemented with technology based on time division multiple access (TDMA) is defined as a time slot of a radio frequency channel. The radio frequency channel can be divided into either three full-rate or six half-rate digital traffic channels (time slots). The length of a TDMA frame divided into six time slots is 40 ms, and the length of each time slot is 6.67 ms. A time slot comprises 162 symbols, and each symbol comprises two bits, so there are 324 bits altogether in a time slot, of which 260 bits contain the data. On a full-rate channel a mobile phone uses for both the transmission TX and reception RX two time slots in a frame (40 ms), as shown in FIG. 2a. On a half-rate channel a mobile phone uses only one time slot in a TDMA frame for the transmission TX and one time slot for the reception RX, as shown in FIG. 2b. The transmission time slot TX and reception time slot RX are always followed by an idle time slot IDLE, which there will be four in the case of a half-rate channel.
FIG. 4a shows the structure of the 6.67-ms TX time slot sent by the mobile phone in the US TDMA system, and FIG. 4b shows the structure of the 6.67-ms RX time slot sent to the mobile phone by a base station. The transmission time slot TX sent by the mobile phone includes a guard time G, power ramp up interval R, speech and channel encoded speech information DATA, synchronization word SYNC and base station identifier CDVCC (Coded Digital Voice Colour Code), and in the reception (reception time slots), instead of the guard time G and power ramp up interval R, extra time RSVD (Reserved). As can be seen from the numbers in FIGS. 4a and 4b representing the amount of bits, there are altogether 324 bits in a time slot, of which 260 are data bits (in the case of speech, data bits containing speech). A burst to be transmitted is generated as described above in connection with FIG. 3, whereby speech is first sampled into 1280 bits and digitized with an A/D converter 1, speech encoding into 159 bits is carried out in a speech codec 2 (using e.g. the VSELP algorithm), channel encoding into 260 bits is carried out in a channel codec 3, and the other bits of the time slot are added in the burst generation block 6 to generate the 324-bit transmission burst which is stored in a buffer register.
From the buffer register the burst to be transmitted is taken to a modulator 7 (such as a .pi./4-DQPSK, or Differential Quadrature Phase Shift Keying modulator) which modulates the burst for the transmission. The .pi./4-DQPSK modulation method is a digital modulation method in which the information is contained in phase changes.
A transmitter 8 mixes the modulated burst through one or more intermediate frequencies to the transmission frequency (824 to 849 MHz) and transmits it via an antenna to the radio path. The transmitter 8 is one of three radio frequency blocks RF. A receiver 9 is the first block on the reception side, and it performs functions which are in reverse as compared to the transmitter 8. The third RF block is a synthesizer 10 which generates frequencies.
Operations are carried out in the reception that are reverse to those performed in the transmission. Following the RF receiver 9 and demodulator 11 there comes bit detection 12 using e.g. a channel equalizer, in which the bits are detected from the received samples, or, the transmitted bit sequence is found out. The detection is followed by deinterleaving 13. After deinterleaving, an error burst, which possibly occurred on the transmission path, is converted into individual error bits that can be corrected in channel decoding. So, after deinterleaving, the detected bits are channel decoded 15 and the error sum is checked using a cyclic redundancy check (CRC). If division of received bits by the same polynomial formula that was used in the transmission (encoding) does not result in parity bits identical to the received parity bits, an error has occurred in the information. When an error is detected, the information frame is corrected or rejected. The channel decoding block 15 not only tries to detect errors but also attempts to correct bit errors occurred in the transmission of the burst. After the channel decoding, the 159-bit speech frame contains the transmitted parameters for speech, which the speech decoding block 16 uses to generate a digital speech signal. The digital speech signal is converted to an analog speech signal in a D/A converter 17. The 20-ms speech signal is collected in a kind of buffer, wherefrom it can be taken to the D/A converter that converts the signal for the loudspeaker 18.
The transceiver includes as an essential controlling unit of the mobile station a control unit 19, which controls substantially all blocks 1 to 18 and coordinates their operation and controls timing. The control unit 19 usually includes a microprocessor among other things.
When a mobile phone is switched on, it is first initialized, after which it goes into the idle state to wait for a mobile paging call transmitted on the control channel or for a user-originated call. Having received a paging call the phone responds to it by attempting to set up a connection to the base station. During the call, the base station monitors the connection quality by means of a mobile assisted handover (MAHO) measurement function, with which the base station may instruct the mobile phone to measure the received signal strength indicator (RSSI) on up to 24 different channels. It is characteristic of a cellular system that not all calls can be terminated within the area of one cell. For instance, a car moving on a motorway may pass by several cells during one call. To facilitate uninterrupted calls, a handover system was created. It is based on the idea that with continuous signal strength measurement at various locations of a cell the cellular system (mobile exchange) can sense when a phone, which is having a call, is moving from a cell into another, whereby the call can be switched over from the first cell (first channel) to the second cell (second channel) "on the move", without losing or interrupting the ongoing call. If the base station finds out on the basis of the MAHO measurement that the connection quality has dropped below a predetermined threshold value, it informs the mobile exchange about the fact and hands the call over to another base station (on the basis of the MAHO measurement).
Several factors influence the decision concerning handover; these include the transmission power of the mobile station, the base station serving the mobile station at that moment, and the neighbouring base stations, the quality of the connection in both directions, represented by the bit error rate (BER), etc. On the basis of these different factors the base station centre (BSC) makes a decision about the handover: whether or not a handover is carried out, and if it is, which is the channel/base station to which the call is handed over. When the base station centre makes a decision about the handover, it sends to the mobile station a handover command which includes the necessary information, such as the data about the new channel and time slot.
For reception, the mobile station operates in the handover as follows: it finds in the incoming signal the synchronization word SYNC of its own time slot RX, on the basis of which it is synchronized to a new base station, whereafter it immediately starts normal reception. However, handover takes some time, during which the mobile station only deals with operations related to the handover (such as adaptation to the channel, settling of the synthesizer to the new channel, getting into synchronization) and speech is therefore not processed, whereby the speech coding functions and audio functions (ie. the speech decoder, D/A converter, speaker, and correspondingly, the microphone, A/D converter and speech encoder) are muted for the duration of the handover because there is no speech in either direction but the connection to the base station is cut off. When synchronization has been obtained, the mobile station can start normal transmission. The handover is then completed.
For transmission, the mobile station operates similarly as in reception, ie. first it finds in the signal coming from the base station its own reception time slot RX, on the basis of which it can calculate the moment of its own transmission time slot.
Depending on the mobile system, the handover may take about 30 to 80 ms, which can be heard on the mobile station as a silent pause. In the reception, one attempted solution for this problem has been to echo the so-called sidetone, which is carried from the earpiece to the microphone, back to the earpiece, albeit a little muffled. Then, in the reception, the silent pause is not so conspicuous. However, this method is viable only in the reception. One alternative for removing the silent pause occurring during a handover, or for making it more inconspicuous, is to feed noise to the mobile phone's earpiece during handover. However, the user may find this, too, disturbing.